I dagens samhälle blir underhållningen mer digital, det blir allt mer vanligt att lyssna på en ljudbok istället för att läsa själv. Eftersom spelindustrin är en av de populäraste formerna av underhållning idag ville vi undersöka hur man kan använda sig av audionarrotologi inom spel. Audionarratologi är läran och studien om ljudets funktion inom ett narrativ. Termen är ett paraplybegrepp från narratologi vilket är läran och studien om narrativ. Vi har använt oss audionarratologi för att se hur ljudet kan samverka med ett textbaserat solo-äventyrsspel, eftersom ljudeffekter vanligtvis inte är ett fokus i spel av denna genre. Vi har utvecklat ett textbaserat spel utan en visuell bild med fokus på text och ljud för att kunna utföra vår undersökning om ljudets roll i ett textbaserat spel. Vi har utfört en kvalitativ studie för att ta reda på vad som är det mest optimala sättet att använda sig av ljud för att förstärka narrativet i ett spel. Programmet som användes för utvecklandet av spelet var Twine. I vårt spel får spelare göra olika val för att kunna fortsätta berättelsen, undersökningen utfördes genom att respondenterna fick fylla i ett frågeformulär och svara på vad de tyckte om ljudet i spelet och hur det påverkade narrativet. Med hjälp av svaren har vi sedan dragit slutsatsen att en bra ljudmix och jämna mellanrum mellan ljud och röster stärker narrativet / In today's society where entertainment is becoming more digital, and it is becoming more common to listen to an audiobook instead of reading one yourself. The game media is one of the most popular forms of entertainment in today's society, we want to see how to make use of audionarrotolgy in video games. Audionarratology is the doctrine and study of the function of sound within a story and a narrative, the term is an umbrella concept from narratology which is the doctrine and study of narration. We've used audionarrotolgy to see how the sound can cooperate with a text-based solo adventure story game, because sound effects are usually not a focus in games of this genre. We have developed a text-based game without visual imagery and with a focus on text and sound, in order to perform research on the role of sound in a text-based game. We conducted a qualitative study to find out what the most optimal way to use sound to reinforce the story in a game. The program that was used to develop the game was Twine. The survey was conducted by respondents filling out a questionnaire and answering what they liked about the sound of the game and how it affected the story. With the help of the answers we have then made our own conclusions for how the sound can interact with the story. We came to the conclusion that having a good sound mix and even intervals for sounds and voices, would make the sound reinforce the narrative.

Cette thèse propose une nouvelle architecture Peer-to-Peer pour l’établissement de sessions SIP (Session Initiation Protocol) sur les réseaux ad hoc. SIP est un protocole con¸cu à l’origine sur un modèle centralisé est n’est pas nativement adapté aux réseaux mobiles ad hoc (MANET) en raison de leurs caractéristiques inhérentes de mobilité. Nous avons ciblé nos études sur un mécanisme de lookup distribué Peer-to-Peer (P2P) tolérant aux fautes, même en cas de mobilité des noeuds du réseau. Cette thèse s’articule autour de quatre principales contributions: Nous introduisons le concept de Structured Mesh Overlay Network (SMON) : un overlay P2P sur MANET permettant d’effectuer des lookups de ressources rapides dans un environnement ad hoc. SMON utilise une architecture cross layer design basée sur une Distributed Hash Table (DHT) utilisant directement les informations de routage OLSR. Cette architecture cross layer permet d’optimiser les performances du réseau overlay lors d’un changement de topologie du réseau. La seconde contribution, SIPMON, est un overlay SIP sur réseau SMON. Sa particularité est d’utiliser un DHT pour distribuer les identifiants d’objet SIP dans le réseau overlay SMON. Les expérimentations menées prouvent que cette approche garantit une durée de découverte SIP constante et permet un établissement de session plus rapide entre deux usagers sur réseau ad hoc. SIPMON ne s’applique cependant qu’à un réseau MANET isolé. Notre troisième contribution SIPMON+ permet un interfonctionnement de plusieurs overlays SIPMON connectés à Internet. SIPMON+ unifie donc les overlays de réseau et permet de joindre un client SIP qu’il soit localisé sur un réseau ad hoc ou sur l’internet. De plus, SIPMON+ permet une continuité de service sans couture lors du passage entre un réseau MANET et un réseau d’infrastructure. Notre prototype a démontré que les performances de temps d’établissement d’appel SIPMON+ étaient meilleures que pour l’approche concurrente MANEMO (MANET for Network Mobility). Le scénario d’usage principal est la fourniture de services de communication multimédia d’urgence rapidement déployables en cas de catastrophe majeure. Nous avons développé un prototype SIPMON+ totalement fonctionnel de service de communication P2P multimédia. Ce prototype a été expérimenté en situation réelle de catastrophe. Notre prototype sans infrastructure a donné de biens meilleurs résultats que MANEMO en termes de temps de déploiement, de taux de perte de paquets et de temps d’établissement d’appel. / This work presents a novel Peer to Peer (P2P) framework for Session Initiation Protocol (SIP) on Mobile Ad Hoc Network (MANET). SIP is a client-server model of computing which can introduce a single point of failure problem. P2P SIP addresses this problem by using a distributed implementation based on a P2P paradigm. However, both the traditional SIP and P2P SIP architectures are not suitable for MANETs because they are initially designed for infrastructured networks whose most nodes are static. We focus on distributed P2P resource lookup mechanisms for SIP which can tolerate failures resulting from the node mobility. Our target application is SIP-based multimedia communication in a rapidly deployable disaster emergency network. To achieve our goal, we provide four contributions as follows. The first contribution is a novel P2P lookup architecture based on a concept of P2P overlay network called a Structured Mesh Overlay Network (SMON). This overlay network enables P2P applications to perform fast resource lookups in the MANET environment. SMON utilizes a cross layer design based on the Distributed Hashing Table (DHT) and has direct access to OLSR routing information. Its cross layer design allows optimizing the overlay network performance during the change of network topology. The second contribution is a distributed SIP architecture on MANET providing SIP user location discovery in a P2P manner which tolerates single-point and multiple-point of failures. Our approach extends the traditional SIP user location discovery by utilizing DHT in SMON to distribute SIP object identifiers over SMON. It offers a constant time on SIP user discovery which results in a fast call setup time between two MANET users. From simulation and experiment results, we find that SIPMON provides the lowest call setup delay when compared to the existing broadcast-based approaches. The third contribution is an extended SIPMON supporting several participating MANETs connected to Internet. This extension (SIPMON+) provides seamless mobility support allowing a SIP user to roam from an ad hoc network to an infrastructured network such as Internet without interrupting an ongoing session. We propose a novel OLSR Overlay Network (OON), a single overlay network containing MANET nodes and some nodes on the Internet. These nodes can communicate using the same OLSR routing protocol. Therefore, SIPMON can be automatically extended without modifying SIPMON internal operations. Through our test-bed experiments, we prove that SIPMON+ has better performance in terms of call setup delay and handoff delay than MANET for Network Mobility (MANEMO). The fourth contribution is a proof-of-concept and a prototype of P2P multimedia communication based on SIPMON+ for post disaster recovery missions. We evaluate our prototype and MANEMO-based approaches through experimentation in real disaster situations (Vehicle to Infrastructure scenarios). We found that our prototype outperforms MANEMO-based approaches in terms of call setup delay, packet loss, and deployment time.

With the rising popularity of Voice over IP (VoIP) services, understanding the effects of a global network on Quality of Service is critical for the providers of VoIP applications. This thesis builds on a model that analyzes the round trip time, packet delay jitter, and packet loss between endpoints on an Autonomous System (AS) level, extending it by mapping AS pairs onto an Internet topology. This model is used to produce a mean opinion score estimate. The mapping is introduced to reduce the size of the problem in order to improve computation times and improve accuracy of estimates. The results of testing show that estimating mean opinion score from this model is not desirable. It also shows that the path mapping does not affect accuracy, but does improve computation times as the input data grows in volume.

<p>When using Voice over IP (VoIP) in Wireless Mesh Networks the overhead induced by the IEEE 802.11 PHY and MAC layer accounts for more than 80% of the channel utilization time, while the actual payload only uses 20% of the time. As a consequence, the Voice over IP capacity is very low. To increase the channel utilization efficiency and the capacity several IP packets can be aggregated in one large packet and transmitted at once. This paper presents a new hop-by-hop IP packet aggregation scheme for Wireless Mesh Networks.</p><p>The size of the aggregation packets is a very important performance factor. Too small packets yield poor aggregation efficiency; too large packets are likely to get dropped when the channel quality is poor. Two novel distributed protocols for calculation of the optimum respectively maximum packet size are described. The first protocol assesses network load by counting the arrival rate of routing protocol probe messages and constantly measuring the signal-to-noise ratio of the channel. Thereby the optimum packet size of the current channel condition can be calculated. The second protocol, which is a simplified version of the first one, measures the signal-to-noise ratio and calculates the maximum packet size.</p><p>The latter method is implemented in the ns-2 network simulator. Performance measurements with no aggregation, a fixed maximum packet size and an adaptive maximum packet size are conducted in two different topologies. Simulation results show that packet aggregation can more than double the number of supported VoIP calls in a Wireless Mesh Network. Adaptively determining the maximum packet size is especially useful when the nodes have different distances or the channel quality is very poor. In that case, adaptive aggregation supports twice as many VoIP calls as fixed maximum packet size aggregation.</p>

As multimedia processing and networking technologies, products and services evolve, the number of users communicating, collaborating and entertaining over the IP networks is growing rapidly. With the emergence of pervasive and ubiquitous multimedia services, this proliferation creates an abundant increase in the amount of the Internet backbone traffic. This brings the problem of efficient transmission of real-time and time-sensitive media content to the fore. Effective multimedia services demand appropriate application-specific and media-aware solutions, without which the full benefits of such services will not be realized. Poor approaches often lead to system performance degradations such as unacceptable presentation quality perceived by the users, possible network collapses due to the high-bandwidth nature of the multimedia applications, and poor performance observed by other data-oriented applications due to the unresponsiveness of multimedia flows. From a networking perspective, traditional approaches consider the application data as "sacred" and do not differentiate any part of it from the rest. While this keeps the data-delivery mechanisms, namely, the transport-layer protocols, as plain as possible, it also precludes these mechanisms from interpreting the media content and tailoring their actions according to the importance of the content. Given that this naive approach cannot satisfy the specific needs of each and every one of the today's emerging applications ranging from videotelephony to video-on-demand, from distance education to telemedicine, from remote surveillance to online video gaming, the study of Multimedia Transport Protocols (MMTP) is overdue. An MMTP solution basically integrates the multimedia content information into the responsible data-delivery mechanisms along with the requirements of the invoking application and network characteristics to deliver the highest level of service quality. In other words, an MMTP solution offers a unified environment where all cooperating protocol components interact with each other and make the best use of this collaboration to fulfill their respective duties. The focus of this thesis is on the design and evaluation of a set of end-to-end and system-level MMTP solutions for scalable, reliable, and high quality multimedia services in ever-changing, complex and heterogeneous computing and communication environments.

Mission-critical system typically refers to a project or system for which the success is vital to the mission of the underlying organization. The failure or delayed completion of the tasks in mission-critical systems may cause severe financial loss, even human casualties. For example, failure of an accurate and timely forecast of Hurricane Rita in September 2005 caused enormous financial loss and several deaths. As such, real-time guarantee and reliability have always been two key foci of mission-critical system design. Many factors affect real-time guarantee and reliability. From the software design perspective, which is the focus of this paper, three aspects are most important. The first of these is how to design a single application to effectively support real-time requirement and improve reliability, the second is how to integrate different applications in a cluster environment to guarantee real-time requirement and improve reliability, and the third is how to effectively coordinate distributed applications to support real-time requirements and improve reliability. Following these three aspects, this dissertation proposes and implements three novel methodologies: real-time component based single node application development, real-time workflow-based cluster application integration, and real-time distributed admission control. For ease of understanding, we introduce these three methodologies and implementations in three real-world mission-critical application systems: single node mission-critical system, cluster environment mission-critical system, and wide-area network mission-critical system. We study full-scale design and implementation of these mission-critical systems, more specifically: 1) For the single node system, we introduce a real-time component based application model, a novel design methodology, and based on the model and methodology, we implement a real-time component based Enterprise JavaBean (EJB) System. Through component based design, efficient resource management and scheduling, we show that our model and design methodology can effectively improve system reliability and guarantee real-time requirement. 2) For the system in a cluster environment, we introduce a new application model, a real-time workflow-based application integration methodology, and based on the model and methodology, we implement a data center management system for the Southeastern Universities Research Association (SURA) project. We show that our methodology can greatly simplify the design of such a system and make it easier to meet deadline requirements, while improving system reliability through the reuse of fully tested legacy models. 3) For the system in a wide area network, we narrow our focus to a representative VoIP system and introduce a general distributed real-time VoIP system model, a novel system design methodology, and an implementation. We show that with our new model and architectural design mechanism, we can provide effective real-time requirement for Voice over Internet Protocol (VoIP).

Over the past decade, the rapid developments and growth of information and communication technology (ICT) have triggered a new wave of customer service. This study utilizes the net-enable business innovation cycle theory with secondary data analysis to analyze the process and outcomes of the implementation of emerging technology, i.e., web-based application and voice over internet protocol (VoIP) for the case company, i.e., A Company. We investigate the characteristics and feature of the emerging technology including the web-based application and VoIP, identify the potential economic opportunity for the A Company, analyze the needed business innovation for its growth, and assess the potential value for its customer. The findings have the potential to contribute to the understanding of impacts occurring in the innovation associated with the implementation of the emerging technology for the A Company and offer rich insights for the company to exploit the economic opportunities, the needed business innovation, and the potential value for the customer. This approach also provides a systematic template that helps an organization to decide whether an emerging technology is worthy to implement.

Die vorliegende Arbeit beschreibt SIP Clients für Linux und Windows. Die Programme funktionieren in einem leistungsstarken Netzwerk, wie z.B. dem Intranet der Technischen Universität Chemnitz, problemlos. Alle Funktionen wurden in einer VoIP-Umgebung getestet. Die Sprachübertragungqualität über das Internet mittels dieser Clients ist jedoch nicht Gegenstand der Arbeit und somit auch nicht erprobt worden. In vielen Unternehmensbereichen fällt häufig der Begriff Echtzeitkommunikation in Verbindung mit einer geeigneten Infrastruktur, ausgelöst durch den Zuwachs der verfügbaren Netzwerkbandbreite und somit immer realistischer werdender Sprach- und Videoübertragungen. Ein Ansatz für die Anwendungen ist das Protokoll SIP (Session Initiation Protocol), welches für die Signalisierung der Video- und Sprachübertragung verwendet wird.

Das Ziel dieser Arbeit ist es, ein Handover-Verfahren für ein Siemens Handset zu entwickeln. Die Entwicklungsumgebung beruht auf den Wireless-LAN Standards 802.11 der IEEE (Institute of Electrical and Electronics Engineers). Dabei liegen die Schwerpunkte auf den Standardisierungen 802.11f und 802.11i, wobei sich eine neue Arbeitsgruppe (IEEE 802.11r) direkt mit dem Thema "Handover" beschäftigen wird. Das Handset soll selbständig die Verwaltung und Einleitung des Handovers übernehmen und lediglich insofern vom Access Point unterstützt werden, dass dieser als Informationssammler dient und somit Entscheidungshilfen geben kann.

Eine kurze Einführung zu "Voice over IP" (dem Telefonieren über Datennetze). Es wird ein Überblick über technische Anforderungen und Lösungen geben. Behandelte Gebiete sind Audio-Codecs, das Transportprotokoll RTP sowie die Signalisierungsdienste SIP und H.323.